The present disclosure relates to the field of telecommunications, and specifically to the field of telecommunication that utilize Voice over Internet Protocol (VoIP). Still more specifically, the present disclosure relates to the field of improving call quality for calls made during a VoIP session.
Voice over Internet Protocol (VoIP) applications have gained wide acceptance by general Internet users and are increasingly important in the enterprise communications sector. However, achieving voice quality levels for VoIP remains a significant challenge, as IP networks typically do not guarantee delay, packet loss, jitter and bandwidth levels. In a VoIP application, voice is digitized and packetized at the sender before its transmission over the IP network to the receiver. At the receiver the packets are decoded and played out to the listener. The process of converting an analogue voice signal to digital is done by an audio “codec”.
A codec is a coder-decoder (hence the name “codec”) that converts an audio signal (e.g., a user's voice detected by a microphone) into a compressed digital form for transmission over a network during a VoIP session at a sending node (e.g., a computer-based telephone). A same type of codec at a receiving node receives the compressed digital transmission signal and converts it back into an uncompressed audio signal for replay at the receiving node. VoIP sessions may be between only two nodes (having only two parties to the VoIP session) or multi-receiver nodes (having multiple nodes receiving a same VoIP audio signal from a sending node).